1. Field
Apparatuses and methods consistent with embodiments relate generally to synchronization of multimedia data, and more particularly, to synchronization of multimedia data by adjusting video playback when a mobile terminal in a wireless communication network receives multimedia data such as audio and video data and plays back the multimedia data.
2. Description of the Related Art
Generally, a server transmits multimedia data such as audio or video playback data to a mobile terminal using real-time transport protocol (RTP)/real-time transport control protocol (RTCP). The mobile terminal adjusts audio and video synchronization by comparing audio and video time stamps contained in the multimedia data with its own system time.
FIG. 1 is a diagram showing audio and video time stamps generated in a server according to a related art method. Referring to FIG. 1, a server generates time stamps with synchronized audio and video. The time stamps are divided into an audio time stamp and a video time stamp. Audio and video sampling time information obtained based on a system clock of the server is included in the audio and video time stamps, respectively.
The audio and video time stamps including the obtained audio and video sampling time information are transmitted to a mobile terminal. The mobile terminal calculates audio and video playback times by comparing the audio and video time stamps received from the server with its own system clock and plays back audio and video at the respective playback times.
However, such a method may cause an unexpected problem such as audio discontinuities or audio skipping when system clock resolution or audio playback rate of the mobile terminal is not identical to system clock resolution or audio playback rate of the server.
For example, audio frames received by a receiver of the mobile terminal should be transmitted to a playback buffer at a transmission rate which is the same as an audio playback rate. If the transmission rate of the audio frames is lower than the audio playback rate, buffer underflow occurs and thereby an audio discontinuity phenomenon occurs. On the contrary, if the transmission rate of the audio frame is higher than the audio playback rate, the buffer is short of space to store data as time elapses and thereby an audio skipping phenomenon occurs. Although such a problem may be overcome in streaming during an interval less than a prescribed time when a large buffer space is assigned for storage of the audio frames, it is difficult to ensure a large buffer space in an environment of a mobile terminal with limited storage capacity.
FIG. 2 is a diagram illustrating audio and video time stamps received by a mobile terminal from a server according to a related method. Due to a problem in which system clock resolution or audio playback rate of a mobile terminal are not accurately identical to that of a server, audio may be played later than video, or vice versa during playback of audio and video. If such a problem is accumulated, audio and video go out of synchronization.
Typically, if there is slight synchronization mismatch and playback is not performed for a long time (generally greater than three minutes), an audio skipping or discontinuity phenomenon may not occur. However, if playback is performed for a long time in a stable streaming environment, the above-described method may generate audio skipping or discontinuities and audio and video synchronization may not be accurately established. Furthermore, since a usable memory resource of the mobile terminal is limited, a method for adjusting audio and video synchronization using a memory less than a prescribed level is needed.